Re: [Asterisk-Users] recommandation for four (4) port FXS ATA
1104 from Mediatrix work very well with asterisk Just becareful to setup correctly REALM and all work like a charm. Mediatrix also have syslog debug,monitoring facility ... I have test some china 4*FXS but i haven't enough time for hard testing. Anyway Mediatrix is not so much expensinve. At 04:04 17/04/2005, you wrote: Can anybody recommend relabel four port FXS ATA adapter? Or any independent reviews. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arnaud Pignard ([EMAIL PROTECTED]) Standard : + 33 1 70 71 50 00 - Fax : +33 1 70 71 50 60 MSN : [EMAIL PROTECTED] - ICQ : 20946060 Frontier Online - Opérateur Internet - http://www.frontier.fr Direct Centrex - Opérateur Télécom sur IP - http://www.directcentrex.com Direct Nom - Registrar de nom de domaine - http://www.directcnom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime UPDATE
Got the same problem, and i rollback to older version (before RT cache patch). There is a combinaison where you will get successfull update but for me, realtime was unstable. I haven't yet time to make more test for maybe report a potential bug. Try modify theses params to yes or no : Rtcachefriends=yes Rtnoupdate=yes Rtautoclear=yes At 01:39 07/04/2005, you wrote: I had discovered this myself. Once I set this value, the updates started to occur, but as shown in my earlier post, are all NULL values. - Original Message - From: Thierry Wehr [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, April 07, 2005 9:36 AM Subject: RE: [Asterisk-Users] Realtime UPDATE -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rod Bacon Envoyé : jeudi 7 avril 2005 01:06 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Realtime UPDATE My problem is that upon registration, the UA's IP address and port information isn't being written to the MYSQL realtime database. Subsequently, calls to the UA fail if they originate from another * server (The server DOES attempt a lookup, but obviously gets no value for IP address / PORT). Hi I'd the same problem and discovered that you have to put rtnoupdate=no in you'r sip.conf Hope it helps you Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
At 15:36 04/04/2005, you wrote: On 03-Apr-2005, Tim Pushor wrote: I prefer PF's approach to security first, convenience second, and I *really* like the fact that PF has a real parser. As the requements get more complex, having everything in one file, and very readable and structured is a huge plus. Also, the integration with ALTQ is nice, especially for these types of applications. I agree with everything Tim wrote above, and I'll add that the biggest factor that influenced me in my move to OpenBSD for my firewall was that it was the only free unix I found that could do bidirectional filtering in bridged mode. As in, when you're in a bridged configuration you can filter in and out on an interface. Neither Linux nor FreeBSD could do this. It's certainly an edge case, but if you need that feature it's invaluable. I'm using ALTQ since FreeBSD 4.6 and it's also exist ALTQ+PF that's near the same as OpenBSD version. And i confirm that's shapping with ALTQ work great ! Even with 32 Kbps. You can easely shape around 1000 rules and have a full Fast Ethernet port on a dual PIII (FreeBSD ALTQ port without PF) ALTQ have many shape algo, maybe the only one with such diversity. You have some CD distribution with ALTQ enable. I posted my asterisk altq experiments here: http://slacker.com/~nugget/asterisk4.php -- David McNett [EMAIL PROTECTED] http://slacker.com/~nugget/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Client
Look here : http://www.voip-info.org/tiki-index.php?page=IAXClient Regards, At 18:22 01/02/2005, you wrote: Hi All, I'd like to develop an IAX - client. Does somebody know where can I get the source code for an IAX client? Regards César ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk/Sip crash Failed to grab lock
Hi, Since around a week i have one asterisk server how stop responding randomely. CVS HEAD with RealTime engine used. The debug log only write Failed to grab lock, trying again... until i stop Asterisk. No more activity for IAX or SIP channels (no log...). CLI still responding. When i try to stop asterisk (stop now or crtl+C) nothing happen and cli die but asterisk still running. This problem appear around each day. I have notice also many new warning : DEBUG[1608]: Avoiding initial deadlock for 'SIP/1.1.1.1-081f28b8' WARNING[1608]: Avoided initial deadlock for 'SIP/1.1.1.1-081f28b8', 10 retries! 1.1.1.1 = another * Seems change in SIP channels make theses new warning. No deadlock channels or problem like this with cvs from mid-december. Any idea ? -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SIP peers in MySQL Database
At 13:53 11/10/2004, Roy Sigurd Karlsbakk wrote: it's in there in -r v1-0, but replaced by some realtime stuff in development CVS I haven't found out more about that, though.. Old mysqlfriends is now remove from asterisk. Now you have to use res_config_odbc for setup sip/iax friends. you can read wiki and this file README.extconfig in docs for get more information how to setup it. You will find also example in extconfig.conf.sample Somebody seems start a mysql drivers for realtime external configuration instead of ODBC. -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: SIP peers in MySQL Database
Yes but it's will be better to have mysql driver At 14:20 11/10/2004, you wrote: Somebody seems start a mysql drivers for realtime external configuration instead of ODBC. You can speak to MySQL with ODBC. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SIP peers in MySQL Database
look at unixODBC or iodbc for more information
Also the reason (i guess) why they move to ODBC is that's ODBC have many
connector to most SQL database.
At 15:26 11/10/2004, you wrote:
Hi all,
Just two questions:
Why asterisk use ODBC(Microsoft?) to connect to SQL
database?
Anybody could answer to my first question ?
Harry
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a écrit
:
res_config_odbc and ast_data is the new way
the old way is still in 1.0.1 and CVS -r v1-0
ast_data is available at
http://svn.asteriskdocs.org/res_data/
roy
On Oct 11, 2004, at 14:47, harry gaillac wrote:
Sorry
I have not look at CVS but I would like somebody
help
me too about my problem.
help please
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a
écrit
:
it's in there in -r v1-0, but replaced by some
realtime stuff in
development CVS
I haven't found out more about that, though..
On Oct 11, 2004, at 13:36, Tomica Crnek wrote:
From few days ago there is no USE_MYSQL_FRIENDS
in channels/Makefile.
That is why I am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 12:19 PM
To: Asterisk Users Mailing List -
Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Re: SIP peers in
MySQL Database
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically
loaded) # #
Copyright (C) 1999, Mark Spencer # # Mark
Spencer
[EMAIL PROTECTED] # # Edited By
Belgarath Aug
28 2004 # Added bare bones ultrasparc-linux
support.
#
# This program is free software, distributed
under the terms
of # the GNU General Public License #
OSARCH=$(shell uname -s)
PROC=$(shell uname -m)
USE_MYSQL_FRIENDS=0
USE_SIP_MYSQL_FRIENDS=0
--- Tomica Crnek [EMAIL PROTECTED] a
écrit
:
It says To enable this, you need to edit the
Makefile in
the channels
directory of your source tree and enable
MYSQL_FRIENDS.,
but there is
no MYSQL_FRIENDS in channels/Makefile any
more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 11:45 AM
To: Glynn Condez
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SIP peers in
MySQL
Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of
i'm
waiting a
reply in order to use sql database for all
sip
clients from
small offices asterisk box with nat context.
May I use autocreatepeer in all asterisk
sip.conf
file with
nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost=
Hostname of server dbuser= Username in MySQL
dbpass= Password
for user in MySQL autocreatepeer=yes nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall
box|--Asterisk
--
+
|
SIP
peers
in
|
mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a
écrit
:
Hi Harry,
how did you make sip peers on mysql
database?
is
it working well?
where can I find a documentation so I could
migrate my Asterisk sip
config to use Mysql also.
Regards
Vous manquez d'espace pour stocker vos mails
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RE: [Asterisk-Users] Re: SIP peers in MySQL Database
No , i use unixODBC on several application/servers. but as you said : 4. It's not much slower than native DB drivers. (15-33% slower) I have never done any bench about it. So i can't make any argumentation on it and seems you have done some bench. However add unixODBC on the middle won't be faster. Let's see future usage realtime external, and imagine all configuration (extension ...) in database on busy server. I would prefer have native mysql driver to reduce load than unixODBC. For most asterisk installation, i agree, unixODBC will fit perfectly. At 15:45 11/10/2004, you wrote: You must be one of those people that doesn't know much about ODBC and is under the impression it's SLOW! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Arnaud Pignard Sent: Monday, October 11, 2004 8:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database Yes but it's will be better to have mysql driver At 14:20 11/10/2004, you wrote: Somebody seems start a mysql drivers for realtime external configuration instead of ODBC. You can speak to MySQL with ODBC. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM40B hangup on fax or data modem carrier
Hi ! I have a TDM40B and i try to use it connected to modem for incoming call data transfert. I have no problem to use it with a phone and a talk communication work fine. But when we try to use with modem, with most modem, we got data carrier for few seconds and channel hungup. [ TYPE: Null Frame (4) SUBCLASS: N/A (3) ] [Zap/4-1] -- Zap/4-1 is ringing [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/4-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/4-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/4-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/4-1] [ TYPE: Null Frame (4) SUBCLASS: N/A (4) ] [Zap/4-1] -- Zap/4-1 answered SIP/213.161.193.64-08142788 [ HANGUP (NULL) ] [Zap/4-1] -- Hungup 'Zap/4-1' I try to configure channel in different mode, without echo cancellation and seems same problem. This configuration is working perfectly with HandyTone from GrandStream. Our zapatel.conf look like this (but i have also test with light configuration) : signalling=fxo_ls ; try also ks ... group=1 relaxdtmf=yes ; make no difference context=sip echocancel=no faxdetect=no; make no difference channel = 1-4 All call are incoming call (from PSTN or SIP G711 - ASTERISK - TDM40B - MODEM) Modem said no carrier when answering with ATA. With a fax machine, i think we will get same problem haven't yet test) Any idea ? Thanks for help ! -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P - ZT_CHANCONFIG failed
Hi, I try setup a TE410P. Already setup E100P without problem. I also check sample zaptel.conf config in mailing list and seems my config is ok. However when i modprobe wct4xxp, here is error output : ZT_CHANCONFIG failed on channel 97: No such device or address (6) FATAL: Error running install command for wct4xxp Here is log and conf file bellow. Thanks for help PS : Debian 2.6.7 --- /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone=fr defaultzone=fr --- /var/log/messages Aug 21 06:36:27 localhost kernel: Found TE410P at base address fe30, remapped to f99dc000 Aug 21 06:36:27 localhost kernel: TE410P version c01a009b Aug 21 06:36:27 localhost kernel: FALC version: 0005, Board ID: 00 Aug 21 06:36:27 localhost kernel: Reg 0: 0x30416800 Aug 21 06:36:27 localhost kernel: Reg 1: 0x30416000 Aug 21 06:36:27 localhost kernel: Reg 2: 0x07fc07fc Aug 21 06:36:27 localhost kernel: Reg 3: 0x Aug 21 06:36:27 localhost kernel: Reg 4: 0x Aug 21 06:36:27 localhost kernel: Reg 5: 0x Aug 21 06:36:27 localhost kernel: Reg 6: 0xc01a009b Aug 21 06:36:27 localhost kernel: Reg 7: 0x1000 Aug 21 06:36:27 localhost kernel: Reg 8: 0x Aug 21 06:36:27 localhost kernel: Reg 9: 0x00ff Aug 21 06:36:27 localhost kernel: Reg 10: 0x Aug 21 06:36:27 localhost kernel: TE410P: Launching card: 0 Aug 21 06:36:27 localhost kernel: TE410P: Setting up global serial parameters Aug 21 06:36:27 localhost kernel: Found a Wildcard: Wildcard TE410P-Xilinx --- cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 796280012 0 0 0IO-APIC-edge timer 2: 0 0 0 0 XT-PIC cascade 4: 7 0 0 0IO-APIC-edge serial 8: 4 0 0 0IO-APIC-edge rtc 14: 22 1 0 0IO-APIC-edge ide0 177: 35086 0 0 0 IO-APIC-level dpti0 185: 767671 0 0 0 IO-APIC-level eth0 201: 42956572 0 0 0 IO-APIC-level t4xxp -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P - ZT_CHANCONFIG failed
Ok, i haven't get in hand the card and make remote hardware install. It's certainely the problem. Thanks ! At 22:49 20/08/2004, you wrote: On Fri, 20 Aug 2004, Arnaud Pignard wrote: I try setup a TE410P. Already setup E100P without problem. I also check sample zaptel.conf config in mailing list and seems my config is ok. However when i modprobe wct4xxp, here is error output : ZT_CHANCONFIG failed on channel 97: No such device or address (6) FATAL: Error running install command for wct4xxp Have you configured the spans for E1 signalling? It sound like you have them set for T1 signalling (24*4=96). You should check the jumpers on the card. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GET VARIABLE with AGI
Hi, Is there a way to get variable as DIALEDTIME or DATETIME ... with GET VARIABLE ? All my test always return unset variable. Else i can pass all variable need by args, but i would prefer more logic way to do it. Regards, -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 codec G7231A6K3
Hi, I would like use codec G7231A6K3 with oh323, but seems asterisk don't undestood this codec. I can't use G7231, the remote gateway don't accept this version of G723. Thanks for help. -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR and EXTEN
For make outgoing call, i setup 0. However 0 is write in the cdr dst field.
Is there a way to remove it when asterisk send it to cdr_mysql ?
exten = _0X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
I just want have in cdr dst = ${EXTEN:1}
This don't work :
exten = _0X.,1,SetVar(EXTEN=${EXTEN:1})
exten = _0X.,2,Dial,SIP/[EMAIL PROTECTED]
Use another variable still record ${EXTEN}
--
Arnaud Pignard ([EMAIL PROTECTED])
Frontier Online - Opérateur Internet
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[Asterisk-Users] cdr and edit dst field
For make outgoing call, i setup 0. However 0 is write in the cdr dst field.
Is there a way to remove it when asterisk send it to cdr_mysql ?
exten = _0X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
I just want have in cdr dst = ${EXTEN:1}
This don't work :
exten = _0X.,1,SetVar(EXTEN=${EXTEN:1})
exten = _0X.,2,Dial,SIP/[EMAIL PROTECTED]
Use another variable still record ${EXTEN}
--
Arnaud Pignard ([EMAIL PROTECTED])
Frontier Online - Opérateur Internet
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[Asterisk-Users] sip_reg_timeout problem
Hello, We have one of our SIP provider that's is sending incoming sip call without need of registration. Incoming call working fine (as outgoing call), but * still try to register to there sip gateway : chan_sip.c:3159 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again -- Got SIP response 404 User Not Found in data base back from 50.50.50.50 Each incoming call is like a new sip user. Auth is made on ip access list (no user / password need). Try to qualify=no and other param and still try Registration. All is working just fine, except this problem registration timeout error. -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip_reg_timeout problem
Hello, Problem was fix by stop asterisk and restart it. Maybe a ghost register in a past configuration file ! At 23:18 26/05/2004, you wrote: Hello, We have one of our SIP provider that's is sending incoming sip call without need of registration. Incoming call working fine (as outgoing call), but * still try to register to there sip gateway : chan_sip.c:3159 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again -- Got SIP response 404 User Not Found in data base back from 50.50.50.50 Each incoming call is like a new sip user. Auth is made on ip access list (no user / password need). Try to qualify=no and other param and still try Registration. All is working just fine, except this problem registration timeout error. -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trunk with CIRPAK
Hello,
I have fix the problem, i haven't notice that's in general i have
videosupport=yes
with this in sip.conf, it's doesn't disable videosupport :
[provider]
host=x.x.x.x
type=peer
videosupport=no
silenceSuppression=no
Now working with videosupport=no in general
At 17:08 07/05/2004, you wrote:
Hello,
I have trouble to enable a sip trunk with a CIRPAK.
CIRPAK support answer that's there parameter are unvalid :
a=silenceSupp:off - - - -
is not standard and not working with cirpak - to be remove
m=video 13072 RTP/AVP
no video, how to remove it ?
my extension.conf :
exten = _6X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
Regards,
--
Arnaud Pignard ([EMAIL PROTECTED])
Frontier Online - Opérateur Internet
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Re: [Asterisk-Users] Problem With zaphfc
rc19 work better for me rc20a is less stable on my configuration (driver crash / line 50% not correctly hangup) At 15:37 23/04/2004, you wrote: Yes i use this version Thank's Tiziano - Original Message - From: mailto:[EMAIL PROTECTED]Robinson Tim-W10277 To: mailto:'[EMAIL PROTECTED]''[EMAIL PROTECTED]' Sent: Friday, April 23, 2004 2:59 PM Subject: RE: [Asterisk-Users] Problem With zaphfc You don't say which version you are using, but upgrade to RC20a. There were some ISDN Layer 2 issues in earlier versions which have been fixed recently. http://ns1.jnetdns.de/jn/relaunch/asterisk/downloads/bri-stuff-0.0.2rc20a.tar.gzhttp://ns1.jnetdns.de/jn/relaunch/asterisk/downloads/bri-stuff-0.0.2rc20a.tar.gz Rgds Tim -Original Message- From: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tiziano Crescimbeni Sent: 23 April 2004 11:42 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem With zaphfc I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:02 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:03 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:25:09 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 2 now, updating n_r! Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 3, but we have nothing -- resetting! Apr 23 12:25:23 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:26 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:39 WARNING[131081]: Ring requested on channel 1 already in use on span 1. Hanging up owner. Apr 23 12:26:22 WARNING[131081]: Ring requested on channel 2 already in use on span 1. Hanging up owner. Apr 23 12:47:33 WARNING[131081]: PRI: Double assgined TEI! Apr 23 12:47:33 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 2 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 3 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 4 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 5 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 6 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 7 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 8 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 9 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 10 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 11 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 12 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 13 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 14 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 15 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 16 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 17 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 18 now, updating n_r! Apr 23 12:47:43 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:47:44 WARNING[131081]: PRI: ACK received outside of window, restarting Apr 23 12:48:16 WARNING[16384]: MySQL database sock file not specified. Using default Apr 23 12:48:16 WARNING[16384]: No '=' (equal sign) in line 34 of mgcp.conf Apr 23 12:48:16 WARNING[16384]: Ignoring port for now Apr 23 12:49:14 NOTICE[311316]: Unable to create channel of type 'Zap' Apr 23 12:49:24 WARNING[311316]: Timeout, but no rule 't' in context 'archimedia' Apr 23 12:49:38 NOTICE[327700]: Unable to create channel of type 'Zap' Apr 23 12:49:48 WARNING[327700]: Timeout, but no rule 't' in context 'archimedia' Apr 23 12:51:39 WARNING[16384]: MySQL database sock file not specified. Using default Apr 23 12:51:39 WARNING[16384]: No '=' (equal sign) in line 34 of mgcp.conf Apr 23 12:51:40 WARNING[16384]: Ignoring port for now
Re: [Asterisk-Users] zaphfc
Try with : channel = 1-2 Regards, At 11:40 20/04/2004, you wrote: Hello, Here it goes: zaptel.conf: --- span=1,1,3,ccs,ami bchan=1-2 dchan=3 --- zapata.conf --- switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local echocancel=yes immediate=yes group = 1 context=local channel = 1 - Thanks, --- Paulo Loureiro. On Mon, 2004-04-19 at 21:27, Arnaud Pignard wrote: Hello, Can you post zapata.conf and zaptel.conf ? It's seems a config file problem. At 19:32 19/04/2004, you wrote: Hello list, I'm trying to use zaphfc, the module loads ok, and it identifies the hfc boards in the machine. The problem is: whenever i try to ztcfg -vv I get the following: 8x--- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) 8x-- when I try to start * it bails out with: == Parsing '/etc/asterisk/zapata.conf': Found Apr 19 17:27:34 WARNING[16384]: chan_zap.c:671 zt_open: Unable to specify channel 1: No such device or address Apr 19 17:27:34 ERROR[16384]: chan_zap.c:5338 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 19 17:27:34 ERROR[16384]: chan_zap.c:7490 setup_zap: Unable to register channel '1' Apr 19 17:27:34 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Apr 19 17:27:34 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Junk at the beginning 49443303 Can anyone out there using zaphfc, help me on this? Thanks in advance, --- Paulo Loureiro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaphfc
You can do something like :
[incoming]
exten = s,1,Answer
exten = s,2,SetCallerID(0${CALLERID})
enten = s,3,
There is maybe a better way to do the samething.
At 18:40 23/04/2004, you wrote:
How i can obtain a complete caller ID from ISDN zaphfc in italy
because i obtain a caller id without a initial 0 (for example
cid=305001010 the correct number is 0305001010)
Thank's Tiziano
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Re: [Asterisk-Users] zaphfc
Hello, Can you post zapata.conf and zaptel.conf ? It's seems a config file problem. At 19:32 19/04/2004, you wrote: Hello list, I'm trying to use zaphfc, the module loads ok, and it identifies the hfc boards in the machine. The problem is: whenever i try to ztcfg -vv I get the following: 8x--- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) 8x-- when I try to start * it bails out with: == Parsing '/etc/asterisk/zapata.conf': Found Apr 19 17:27:34 WARNING[16384]: chan_zap.c:671 zt_open: Unable to specify channel 1: No such device or address Apr 19 17:27:34 ERROR[16384]: chan_zap.c:5338 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 19 17:27:34 ERROR[16384]: chan_zap.c:7490 setup_zap: Unable to register channel '1' Apr 19 17:27:34 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Apr 19 17:27:34 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Junk at the beginning 49443303 Can anyone out there using zaphfc, help me on this? Thanks in advance, --- Paulo Loureiro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc problem
Hi,
I have a partial working installation with zaphfc.
Incoming call :
For incoming call, seems work fine. But the sound is very bad with bounce
short crashing sound. Same sound with echo cancel off or on.
SDA work fine.
Another problem, it's seems that's zaphfc don't reset correctly the line. I
have one of my D channel how was busy even after stop communication.
Outgoing call :
When try make a call, i have error like this :
Mar 18 22:44:05 WARNING[229391]: chan_zap.c:5952 zt_pri_error: PRI: !! Got
reject for frame 1, but we have nothing -- resetting!
MFE for TEI = 80
== D-Channel on span 1 up
== D-Channel on span 1 down
== D-Channel on span 1 down
Config is mostly like howto on voip-info.org
in /var/log/messages, i have hundred of this line :
zaphfc: empty HDLC frame received
---
Hardware : Bewan Gazel PCI (have his dedicaced IRQ)
---
ztcfg :
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
---
/etc/zaptel.conf :
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
fxsks=4
---
/etc/asterisk/zapata.conf :
[snip]
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan=local
echocancel=yes
immediate=yes
;setcallerid(${CALLERIDNUM})
;usecallerid=yes
group = 1
context=incoming
channel = 1-2
[snip]
Don't work with bri_net_ptmp
---
ISDN operator : France Telecom
---
*CLI zap show channel 1
Channel: 1
File Descriptor: 25
Span: 1
Extension: s
Context: incoming
Caller ID string: xx
Destroy: 0
Signalling Type: PRI Signalling
Owner: Zap/1-1
Real: Zap/1-1
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 0 taps unless TDM bridged, currently OFF
PRI Flags: Call
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
When offline :
[snip]
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
PRI Flags:
Mar 18 22:40:57 WARNING[16384]: chan_zap.c:7055 zap_show_channel: Failed to
get conference info on channel 1
Mar 18 22:40:57 WARNING[16384]: chan_zap.c:7061 zap_show_channel: Failed to
get confmute info on channel 1
Thanks for help !
--
Arnaud Pignard ([EMAIL PROTECTED])
Frontier Online - Opérateur Internet
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